1. Field of the Invention
The present invention relates to a method for improving the quality of packet switched transmissions, and more particularly to a notification system that informs either the end user or control path system of transmission impairment issues in a packet switched network. The present invention more particularly relates to the field of IP telephony, and especially to media quality control systems that actively mitigate or eliminate sources of impairment to real-time (voice and video conferencing) media flows.
2. Description of the Relevant Art
FIG. 1 depicts a schematic diagram of a conventional telecommunications system 100. System 100 comprises telephones 101 and 109, Public Switched Telephone Networks (PSTN) subnetworks 102 and 108, Internet Protocol (IP) gateways 103 and 107, Internet Protocol network 104, and Internet Protocol endpoints 105 and 106, interconnected as shown. System 100 enables telephones 101 and 109 and endpoints 105 and 106, as well as other telecommunications terminals, to communicate with each other various kinds of media such as audio, video, and so forth. FIG. 1 illustrates an interconnection between PSTN's using IP network interconnectivity. Although not illustrated in FIG. 1, it should also be noted that IP networks using PSTN interconnectivity is common, and that an all-IP network without any PSTN interconnectivity is also possible. The present invention can be directed to the network of FIG. 1, as well as an IP networks using PSTN interconnectivity or an all-IP network without any PSTN connectivity.
Each of telephones 101 and 109 is a telecommunications terminal that is capable of making calls to or receiving calls from any other telecommunications terminal—PSTN-based or IP-based—in telecommunications system 100.
Public Switched Telephone Network subnetworks 102 and 108 are portions of the Public Switched Telephone Network (PSTN). Subnetwork 102 comprises access paths, switches, and transmission paths, in a combination of analog and digital technology, which enable telephone 101 to communicate with other terminals. Subnetwork 108 also comprises access paths, switches, and transmission paths, in a combination of analog and digital technology, which enable telephone 109 to communicate with other terminals. Each depicted portion of the PSTN might comprise wireline equipment, wireless equipment, or both wireline and wireless equipment.
Internet Protocol gateways 103 and 107 are nodes that act as access points into Internet Protocol network 104 for signals from PSTN subnetworks 102 and 108, respectively.
Each of Internet Protocol endpoints 105 and 106 is a packet-capable telecommunications terminal, such as an IP telephone, that communicates via the Internet Protocol. Each endpoint 105, 106 is capable of making calls to or receiving calls from any other telecommunications terminal—PSTN-based or IP-based—in telecommunications system 100.
Internet Protocol network 104 is a packet-switched network that is capable of transporting packets from one node to another. The transported packets can comprise voice or video signal information in their payloads and can also comprise Real-time Transport Protocol (RTP) headers, User Datagram Protocol (UDP) headers, or IP headers. When the packets comprise voice signal information with IP headers, they are often referred to as Voice over Internet Protocol (VoIP) packets, and the networks that transport the VoIP packets are often referred to as VoIP networks.
The service provided by a network path in network 104 can be characterized by its “quality of service,” which, for the purposes of this specification, is defined as a function of the bandwidth, error rate, and latency from one node to another. For the purposes of this specification, the “bandwidth” from one node to another is defined as an indication of the amount of information per unit time that can be transported from the first node to the second. Typically, bandwidth is measured in bits or bytes per second. The bandwidth exhibited by the network can be compared to the bandwidth requirements of one or more media flows; the “bandwidth requirement” is the amount of information per unit time per media flow that has to be transported from the first node to the second, usually determined by the signal encoding protocol (e.g., G.711 for voice, etc.) that governs the particular media flow. For the purposes of this specification, the “error rate” from one node to another is defined as an indication of the amount of information that is corrupted as it travels from the first node to the second. Typically, error rate is measured in bit errors per number of bits transmitted or in packets lost per number of packets transmitted. For the purposes of this specification, the “latency” from one node to another is defined as an indication of how much time is required to transport information from one node to another, plus any packetization delays and buffering delays that accumulate at the endpoints. Typically, latency is measured in milliseconds. The quality of service provided by network 104 can vary based on the actual bandwidth, error rate, and latency experienced by the call or session that is being carried by network 104, in relation to the requirements for the bandwidth, error rate, and latency for the call or session.
The quality experienced in telecommunications system 100 can also depend on other factors. First, each of telephones 101 and 109 can influence audio call clarity through the quality of its loudspeaker and microphone, the loudness of the transmitted and received signal, and the acoustic echo generated between the loudspeaker and microphone. Second, where each of PSTN subnetworks 102 and 108 converts the analog voice signals from a telephone into digital signals to yield greater efficiency in the transmission backbone, digitizing those voice signals can affect the clarity. Third, each of gateways 103 and 107 can affect the clarity through its components such as speech codecs, silence suppression mechanisms, comfort noise generators, jitter buffers, and echo cancellers. And fourth, each of endpoints 105 and 106 also can affect the clarity through its components such as a speech codec, a silence suppression mechanism, and the quality of its loudspeaker and microphone. Many of the impairments that are presently experienced by telecommunications users are as the result of different networks, such as Voice over Internet Protocol networks versus the PSTN, having to interoperate with each other, where some of those networks—or at least the commercial application of those networks—are relatively new in telecommunications. Both the differences between the networks and the equipment that is necessary to enable the different networks to interoperate, such as gateways, are some of the causes of impairments, many of which were either imperceptible or nonexistent in a PSTN-only telecommunications environment.
Other configurations of “hybrid” telecommunications systems that comprise both PSTN and IP-based networks also exist in the art. For example, an IP network using PSTN interconnectivity or an all-IP network without any PSTN connectivity are possible. In those other systems, as in the telecommunications system described above and with respect to FIG. 1, both the PSTN and the gateways that bridge the PSTN and IP-based networks can be sources of impairments.
Compelling business metrics, such as cost, flexibility, and functionality, have driven a migration of real-time distributed applications (e.g., voice and video conferencing) from using circuit-switched or PSTN networks 102, 108 for media transmission to using IP-based packet-switched networks 104 for media transmission. Packet-switched transmission introduces new sources of media quality impairments; however, unlike legacy PSTN networks 102, 108 actions may be taken in a packet-switched network 104 to mitigate or eliminate the cause of impairment. For example, the path for the media flow between two phones 105, 106 may traverse a WAN interconnection that may be temporarily congested, causing noticeable delay and distortion from packet loss; however, the media flow path may be changed to use a non-congested WAN interconnection or rerouted through the Internet and thereby remove the source of impairment. Media quality control technology such as a Converged Network Analyzer (“CNA”) and Inter-Gateway Alternate Router (“IGAR”) by Avaya Inc. are capable of changing the path of media flow.
As used herein, the term “media quality control” means any technique or device for controlling or improving the quality of media transmitted across a network including path control hardware or methods, codecs, or echo cancellation, etc. Media quality control systems which provide path control typically include an impairment detection subsystem that monitors and measures media quality in a (enterprise) network. Typically, when quality measurement values (e.g., estimated MOS) cross thresholds, the subsystem notifies the media quality control subsystem, which then effects some change in the path to bypass the source of impairment. Such systems do not “work perfectly”. There are gaps between the ideal/optimal system and the state-of-the-art system.
Impairment detection and source location is a complex problem, so impairments may occur that are noticeable to the (human) end users but that are not detected by the media quality control system, and therefore no action will be taken by the system to mitigate the problem. This category of gaps can be referred to as the “false negatives” category.
When the media quality control system detects impairments or suspects that impairments may occur, there may be a significant latency between the detection of the problem or potential problem and the action taken to mitigate the problem. During this latency period, at best the end users will be annoyed with the reduced quality of their communications experience, and at worst the end users will abandon the call. This category of gaps can be referred to as the “user feedback” category.
Existing solutions known in the art exhibit false negatives, and users who are annoyed by impairments associated with false negatives may abandon calls. Accordingly, there is a need for a bidirectional notification system that eliminates false negatives and reduces user annoyance and the probability that calls will be abandoned.